This invention relates to IP telephony and, more particularly, to method and apparatus for insuring that IP telephony is carried out in a manner that is effective for customers.
Conventionally, a business enterprise's telephonic interactions with customers, providers, and branches of the enterprise are conducted through a PBX having trunks that connect the PBX trunks to the local public switched network. The local network is connected to an inter-exchange carrier's network, and to the network of international telephone service providers. The business enterprise's computer interactions are typically conducted through a corporate local area network, or wide area network, followed by a digital network (often, via a gateway) which, typically, is a packet network. This may be an Asynchronous Transfer Mode (ATM) network, an IP network, Frame Relay network, or some other network. Through software control a segregated network is sometimes created within the general-use packet network, and such segregated networks are called Virtual Private Networks (VPNs). Business enterprises often purchase such VPNs with a certain load carrying capability.
In recent years the notion of IP telephony has been making inroads into the realm of telephone communications, where telephone calls are digitized, and are carried over the same network as computer interactions. Relative to situations where a customer has resources for data communication and separate resources for voice communication, combining data and voice so that a single resource may be used sometimes has a cost advantage because resources can be utilized more efficiently. Moreover, utilizing the network that was previously dedicated for computer communications has the advantage of bypassing toll charges from inter-exchange carriers and international telephone service providers. In a business enterprise environment, this is achieved by dedicating a number of trunks of the PBX to Internet telephony, and connecting those trunks to a voice-enabled Internet router, for example, a router manufactured by Cisco Systems. The voice-enabled router includes interface circuitry that effects encoding of an analog telephone call to digital to permit transfer over the IP network, and then decoding back to analog at the receiving side. The function is bidirectional, permitting full duplex phone conversations. Since this interface circuitry is conventional, its precise structure is not relevant to the principles disclosed herein and, therefore, it is not further described herein.
FIG. 1 presents a block diagram of such an arrangement, comprising packet network 100 and PSTN 200, which illustratively, is made up of a first domestic transport network, 210, a second domestic transport network, 230, and international transport network 220. Calls between telephone 11 in New York City, for example, and telephone 15 in Sao Paulo, for example, traverse PBX 21 and networks 210, 220, and 230. More specifically, when telephone 11 wishes to establish a call to telephone 15, a process within PBX 21 decides whether the call ought to be made via PSTN 200, and if so, PBX 21 outputs the call on one of the trunks that is connected to switch 211, and thence the conventional call establishment process takes over. That is, the call is extended to switch 211 in network 210, then to international network 220, then to domestic network 230 and to switch 232 within it, and lastly to PBX 25. PBX 25 connects to telephone 15.
When the process within PBX 21 decides that the call ought to be made via packet network 100, it outputs the call on one of the trunks (22) that is connected to interface circuitry 311 in voice-enabled router 31 in, or near, the location of telephone 11, i.e., in, or near, New York City. In accordance with known voice-over-IP (VoIP) techniques a packet flow is established between interface circuitry 311 and interface circuitry 321 within voice-enabled router 32 that is in, or near, Sao Paulo, and interface circuitry 321 output signals on a trunk line 34 that connects to PBX 25.
Generally speaking, a router such as router 32 can have a multiple number of output port groupings, and each grouping is connected to a different PBX and, conversely, a PBX can be connected to a number of routers. FIG. 1 shows two trunk groupings from router 32 (one connected directly to PBX 25, and one connected to switch 231, to demonstrate that connection of a voice-enabled router 32 is not limited to a PBX (such as PBX 25).
In operation, when PBX 21 receives the telephone number of called party 15 and decides to route the call through packet network 100, router 31 identifies the trunk grouping at the output of router 32 that needs to be addressed, and addresses packets to that trunk grouping; i.e., to trunk grouping 34. When the packets arrive at router 32, they are converted to the format acceptable to PBX 25, directed to trunk grouping 34, and applied to PBX 25. The data thus provided to PBX 25 includes the called party's telephone number, and PBX 25 establishes a connection to telephone 15.
If the destination telephone (of the call made by telephone 11) is telephone 12, which is not behind a PBX, as is telephone 15, the principles of this invention still apply. Router 31 identifies that trunk grouping 35, the destination information is provided to switch 32, switch 32 routes the incoming call to switch 231 in domestic transport network 230, and through network 230 to switch 232, which establishes a connection to telephone 12. It is noted that since telephone 12 is not behind a PBX that can make decisions as to whether to complete a call using the PSTN network or the IP network, the operation is not symmetric; that is, calls initiated by telephone 11 can enjoy the benefits of this invention, but calls initiated by telephone 12 cannot.
This asymmetry can be rectified, of course, by embedding the decision-making capacity of PBXs 21 and 25 within switch 232.
It may also be mentioned that some, or all, of the telephones that can gain access to the FIG. 1 routers can be IP telephones, reducing the data conversion burdens of the voice-enabled routers.
One issue, of course, relates to the above-mentioned decision: when ought a call be established via network 100, and when via network 200. Cost and availability of a channel are the primary factors and, hence, it is desirable to provide a means for determining the effectiveness of the hard-wired split, i.e., the number of PBX 21 trunks that are connected to switch 211 and the number of PBX 21 trunks that are connected to interface circuitry 41.